Digital data transfer has been an important technology for many years. Traditionally, digitally transferred data has been packaged as files. When a file is transferred, the loss of data is often critical, because the data contained in the file defines not only the information of the file, which can often be reconstructed, but the structure of the file, which informs the reading device where the file begins and ends, and which data is to be viewed as part of the file. Moreover, digital data transfers have traditionally been accomplished by a server connected more or less directly to a client through a telephone line or similar device. While such a connection is susceptible to losses, the losses are usually not excessive.
Traditional digital data transfer, because of the need to transfer a file with its complete data and structure, and because of the relatively low losses introduced by traditional transfer methods, has typically been intolerant of losses. Typically, data transfers were monitored for losses, and any lost data was retransmitted. The need to retransmit data naturally increased the time required for a data transfer, but the additional time required was usually not great, and the added time was required because the data needed to be transmitted without errors.
More recently, data transfers have increasingly taken place over networks lacking a dedicated connection between server and client, with the leading such network being the Internet. Data is transmitted over the Internet through routings which may differ from the transmission of one packet to the next. This contributes significantly to the likelihood that data will be lost. Moreover, a substantial portion of the data transmitted over the Internet consists of graphics and sounds. An acceptable transmission of such items may be highly tolerant of losses, but less tolerant of delay.
Many Internet users would be only too glad to sacrifice a certain amount of picture quality in order to have a picture display faster. Even more importantly, real-time voice communication is increasing in importance as an Internet application. If data losses occur during a real-time voice communication, the degradation in quality caused by these losses may be imperceptible, while the time lost in retransmitting lost packets in order to reconstruct the data without error would adversely cause an interruption in the conversation. Thus, while in such an environment, a loss-free transmission is ideal and preferable so that no packets would be lost, in a real world transmission in which packets are lost, an imperfect transmission of the data would be highly preferable to the delays which would be occasioned by the retransmission of the lost packets.
There exists, therefore, a need in the art for a way to transmit data by means of digital packets such that a reduced quality reconstruction of the data is possible if a certain number of packets are lost, with a finer reconstruction being achieved with the loss of fewer packets.